I've got the same issue with a Swissvoice IP10S SIP phone. I couldn't
find much information with this issue, but it seems to appear because
Asterisk does not support variable length for g.729 (don't ask me what
it really means). Anyway, it is recommanded to disable the silence
suppression, which seems related to this issue. Unfortunately, it
didn't work for me. But after setting the phone to canreinvite=no in the
sip.conf, the connection worked allright. Don't ask me why.
Jean-Christophe
Bartosz Jozwiak a �crit :
I have this showing on my cli while being in a call.
Then connection gets broken.
Can someone tell me what it means ?
Dropping frame of G.729 since we already have a VAD frame at the end
Thank you in advance.
Bartosz
_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users